22 feb 2007 kl. 12.20 skrev Steve Langstaff:
Are the RTP timers applicable with canreinvite=yes ?
how could we possibly check RTP if the RTP doesn't touch or network
card at all?
The timers are only used when we have RTP streams going to us. If the
RTP stream
is redirected, it's up to
: Re: [asterisk-users] Channels hanging when SIP phone
> gets resetduring call
>
>
> 21 feb 2007 kl. 12.54 skrev Steve Langstaff:
>
> > Hi All.
> >
> > This is on Asterisk 1.2.13
> >
> > I place a call between 2 SIP phones (with canreinvite=yes,
> &
